IRC log for #asterisk on 20110818

00:01.06*** join/#asterisk Cain (~Geek@unaffiliated/cain)
00:06.09*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
00:41.02p3nguinDoes anyone have any idea why asterisk System() is not capable of sending email with mutt anymore?
00:41.13*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
00:43.19ChannelZI can't imagine why it wouldn't
00:46.29p3nguinI've been fucking with that fax stuff all damn evening thinking res_fax or res_fax_digium was where the failure was.
00:46.49*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
00:47.21p3nguinTurns out, asterisk can't mail with mutt anymore.  At least it can't the way I was doing it.  System(echo "stuff to say" | mutt -s "Some subject" recipient)
00:47.40*** join/#asterisk luckman212_phone (~luckman21@2001:470:1f07:1225:c5a3:7ab4:c149:ddcf)
00:51.03ChannelZhmm I've never used mutt as an MTA, just to read archived mbox files
00:51.13p3nguinIt's a client, not an MTA.
00:51.27p3nguinSo the fax was arriving successfully, but it wasn't being emailed.
00:51.35ChannelZUsing it _like_ an MTA
00:51.46ChannelZA mini one as it were.
00:51.48p3nguinI'm using it just like any other command line client.
00:51.57p3nguinIt's a typical email client.
00:52.05p3nguinIt uses the local MTA to send.
00:52.16ChannelZForget it, I'm sick of the nitpicking semantics arguments you so love
00:52.17p3nguinIn my case, it's just msmtpd relaying to gmail.
00:52.20*** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16)
01:01.10pdtpatrickQuestion .. is this bad ?
01:01.28pdtpatrickexten => 6676,1,Dial(IAX2/${SECRET}@int-voip.te-c.com/${EXTEN})
01:01.30ChannelZYes, it's a horror show!
01:03.21WIMPyWhy don't you use a peer?
01:03.55WIMPyAre you trying to fix dundi by not using it?
01:05.33pdtpatrickI;ve created the keys and the boxes have the keys  .. dundi show peers shows all the peers
01:05.37pdtpatrickmappings look fine
01:05.41pdtpatrickkeys show look fine
01:05.46pdtpatrickhowever lookup produces nothing
01:06.09WIMPyCheck the included contexts in dundi.conf.
01:08.22*** join/#asterisk coppice (~chatzilla@116.92.38.165)
01:11.43*** join/#asterisk marits (~gm@c-24-4-226-112.hsd1.ca.comcast.net)
01:18.22*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
01:19.24*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
01:37.11*** join/#asterisk methodvon (~methodvon@pool-71-191-175-251.washdc.fios.verizon.net)
01:44.39*** join/#asterisk sorressean (~tyler@tds-solutions.net)
01:44.55sorresseanI'm looking to set up a system, what sort of service do people use for inbound calls in the US?
01:46.59*** join/#asterisk fireman_biff (~biff@65.48.132.153)
01:48.44*** join/#asterisk dijib (~nobodysho@d72-39-65-1.home1.cgocable.net)
01:49.39dijibgood evening all
01:50.25WIMPy~itsplist-us
01:50.25infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
01:50.30WIMPysorressean: ^^
01:50.59sorresseanWIMPy:  thanks
02:05.10sorresseanWow. they want $20 per channel? Is there a cheaper way about going about this?
02:06.51*** join/#asterisk james_zhu (~Administr@183.16.209.216)
02:09.23dijibvoip.ms is reasonable no?
02:15.12phixwhat?
02:15.59phixi dont like anything with ms in it as i dont have thousands or millions of dispisable income
02:16.21phixdisposible even, touch pad fail
02:22.11fireman_biffAll calls are dropping every 10 - 20 mins or so, with these messages in /var/log/asterisk/full: "Write to 42 failed: Unknown error 500; Short write: 0/15 (Unknown error 500)". I'm using analog phones and a PRI, asterisk 1.4.22
02:23.01fireman_biffand occasionally messages like these also appear at the time of the dropped calls: "Got reject for frame 10, retransmitting frame 10 now, updating n_r! Got reject for frame 11, but we have nothing -- resetting!"
02:23.09fireman_biffany ideas?
02:28.07p3nguinphix: VoIP.ms has nothing to do with multiple sclerosis.
02:31.11WIMPyfireman_biff: Looks like you have severe transmission trouble. Is your timing ok?
02:33.39fireman_biffWIMPy: not sure, how would I check? everything was fine with the box until today
02:34.22WIMPyDo you have rodents in your server room?
02:34.52WIMPyThe timing thing unfortunately only appears in dmesg.
02:35.35WIMPyNo changes since it happens?
02:37.05fireman_biffno changes, and no rodents unless you count usb mice
02:37.09p3nguinWhat would prevent asterisk from being able to run mutt in System()?
02:37.18p3nguinIt can run mutt from the CLI.
02:37.38fireman_biffthe one change since it started was that i disabled echo cancellation after reading that suggestion somewhere, but it had no effect
02:37.59WIMPyEC is not going to change anything.
02:38.16fireman_biffI'm not sure what I'm looking for in dmesg, but I'm not seeing anything that looks like an error
02:38.17WIMPyYou have trouble somewere between your card and your Telco.
02:38.42WIMPyLet's see your dahdi/system.conf
02:38.46*** join/#asterisk corretico (~luis@201.201.44.82)
02:39.22p3nguinI'm sure this worked in an earlier version.  Maybe I need to go back to a really old asterisk, see if it works, and, if it does, start increasing the version until it stops working.
02:39.43fireman_biffi dont have that, would it be zaptel.conf in older versions?
02:40.14WIMPyp3nguin: Why would you start a client from a demon? That's not the obvious choice.
02:40.23WIMPyfireman_biff: yes
02:41.14fireman_biffhttp://pastebin.com/Y2vpJVRc
02:41.51p3nguinI'm just trying to email my fax to my email inbox.  Basically this: System(/bin/echo "See attachment"|/usr/bin/mutt -a my-fax-file -s "New fax" -- email@google.com)
02:42.46p3nguinIt used to work when I was running a different computer.  I changed computers, forgot to fix faxing for a while, finally got around to it, now mutt won't email me anymore.
02:42.54WIMPyfireman_biff: That looks ok. Have you tried to check cabling?
02:43.18p3nguinI can run the full command on the asterisk CLI and it works fine.  Put it in System() and it never works, and I can't find any way to debug it.
02:44.07fireman_biffWIMPy: no, haven't checked that yet, I had been assuming it was something software related
02:44.18p3nguinThe /bin/echo part runs fine from inside System(), so I ruled out that part of it.  mutt is the only thing left.
02:44.38WIMPyfireman_biff: Why would you assume that unless you changed something?
02:45.19fireman_biffWIMPy: most of the issues we've had in the past were solved by a restart, or something on the providers end
02:45.34fireman_biffnever had a problem with the cables connecting to the pbx before
02:45.39fireman_biffso i just didnt think of it
02:45.39p3nguinI just found a post where someone is having a similar problem; he says to use sudo to run mutt from inside asterisk.  Maybe it has something to do with asterisk not having a shell.
02:45.48p3nguinI could give it a shell and try again.
02:45.50p3nguinwithout sudo.
02:46.46WIMPyWe really need a BERT application.
02:47.27p3nguinWell, that didn's fix it.
02:48.21fireman_biffgonna check the cables, be back in a few minutes
02:51.05lkthomashey guys
02:51.07p3nguinSUCCESS
02:51.22lkthomaswe have asterisk connect to ATA then connect to a fax machine
02:51.23p3nguinI don't get it, but using sudo to run mutt from System() worked.
02:51.44lkthomasthe fax machine could send out fax but can't receive
02:52.13lkthomasany special parameter need to set on asterisk or ATA to get fax working ?
02:52.41WIMPylkthomas: A lot. how are you (trying to) send faxes?
02:53.15lkthomasfrom that fax machine
02:53.52WIMPyWe need to know the whole way from one fax machin to the other. Inboth ways, if different.
02:54.15lkthomasPSTN <> Asterisk <> ATA <> fax machine
02:54.19lkthomasincoming and outgoing is the same
02:54.42WIMPyOk, what pstn connection?
02:55.13lkthomasT12
02:55.15lkthomasT1
02:55.26lkthomasT1 connection, sorry for the typo
02:55.49lkthomaswhen we call the fax phone number, it keep ringing until timeout
02:56.13WIMPyFor all numers or only certain ones?
02:56.16lkthomaswe could see fax machine have signal receive call, but it can't pick up
02:56.18*** join/#asterisk seraphie (~erin@75.76.38.159)
02:56.23lkthomascertain one
02:56.28lkthomasonly one number for fax
02:56.45WIMPyhttp://voice.yeti.dk/AvI#t-pit
02:57.25WIMPyJust written a few hours ago :-)
02:58.42lkthomas3k1 is in place
02:59.54WIMPyOk, well, then it might be best to check with the other party, what's going on there.
03:01.37WIMPyErr, wait...
03:02.19WIMPyI was obviousely still in the sending topic. It's about receiving.
03:02.34lkthomaserrrrr
03:02.50WIMPySo what exactly happens when the fax is called?
03:03.22lkthomaskeep ringing
03:03.25lkthomasuntil timeout
03:03.36lkthomasfax machine unable to pick up the call
03:03.48WIMPySo it does ring, but the fax won't answer?
03:03.57WIMPyOr does it answer and nothing happens?
03:04.07lkthomaswon't answer
03:04.53WIMPyHave you tried to connect a phone insted to see it it really rings?
03:05.24lkthomasgood question
03:06.04lkthomaswe don't have analog phone
03:06.07lkthomasso can't test
03:08.43WIMPyYou could try to manually answer on the fax while it should be ringing.
03:09.03lkthomasyou mean pick up the phone on the fax machine ?
03:09.29WIMPyOh, it is with phone? Well, yes, try that.
03:09.51WIMPyIs it configured to receive faxes?
03:10.03lkthomasactually I also think of this
03:10.05*** join/#asterisk radic (~radic@dslb-094-216-229-190.pools.arcor-ip.net)
03:10.10lkthomasit might be because fax machine misconfig
03:10.22lkthomasbut we have search the manual and can't find a setting not to answer incoming fax
03:11.28WIMPyOn these combodevices you ysually have to set the mode to phone only / manual fax, fax only or autoanswer with fax switching.
03:12.52WIMPyIt might even have a dedicated button for that.
03:13.11ChannelZI've got an MFC thingy and you do have to specifically config it to answer after so many rings
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03:26.14lkthomasbrb
03:26.15lkthomasreboot
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04:18.16justdaveok, is it just me, or is it silly that the "to-continue-in-english" sound file is actually translated to french in the french language pack?
04:18.35justdavethat particular prompt I would think you'd still want in english.
04:18.50WIMPyWould make sense to me.
04:19.00justdaveand where do I find the corresponding "to-continue-in-french" said in French to put in the english pack?
04:19.08justdave(it doesn't even have that one in french)
04:19.24WIMPyI probably wouldn;t understand that I could continue if it was told in french.
04:19.41justdaveyou wouldn't need to, that prompt isn't intended for you
04:20.02justdaveit's intended for people who would rather hear the prompt in french. :)
04:20.04WIMPyThat's not a standard sample.
04:20.33WIMPywas referring to the to-continue-in-english one.
04:20.49justdaveoh, right. :)  yeah, that's my point.
04:21.05justdaveThe use case is the call comes in on the french phone line, so the language code is set to french.
04:21.23justdavephone tree leads off with "To continue in English, press 1" then proceeds to give the menu in french.
04:21.40justdavethe To continue in English part should actually be in English
04:21.40WIMPyYou could set the default language depending on the callerID.
04:21.55WIMPyyes
04:23.18justdaveI guess the smart thing to do is just have the receptionist re-record both localizations of the language file with that part included in the main file
04:25.00WIMPyIt never hurts to have multiple small files. You might find them useful elsewhere.
04:25.58*** join/#asterisk snadge (~snadge@unaffiliated/snadge)
04:26.16WIMPyslipt uot the vm-deleted the other day, because I needed a single "deleted".
04:26.52snadgei need some help with iinetphone, im hoping someone has some experience with it.. as its epically retarded, and all the instructions regarding asterisk.. are either outdated, misleading, or just plain wrong
04:27.23snadgei would've given up on this ages ago, except it goes through phases of working fine.. and then not working.. and i can never figure out why
04:28.17*** part/#asterisk fireman_biff (~biff@65.48.132.153)
04:28.30snadgethe sip debug shows a million registrations going out.. and nothing coming back.. sometimes it succeeds.. and i can see state registered.. but then it tries to re-register and fails again, state sent
04:31.01snadgeAsterisk 1.4.42
04:31.41snadgewhen i place an outgoing call.. theres a 10 second delay or so, then it comes back "all circuits are busy, please try again later"
04:37.47*** join/#asterisk dijib (~nobodysho@64.250.95.237)
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04:39.20dijibp3nguin, do you ever stop?
04:40.14*** join/#asterisk james_zhu (~Administr@183.16.209.216)
04:41.24dijibive got an Spawn extension (macro-stdexten, s, 1) exited non-zero on
04:41.29dijiberror if anyone can help
04:41.42*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
04:41.42dijibdrops calls.
04:42.00WIMPyIn Asterisk land, we don't sleep(), we wait().
04:42.00ChannelZPerhaps we'll conjure a solution
04:42.07dijiblol
04:42.18WIMPyLooks like someone hung up
04:42.35dijibonly incomming calls, and no party hangs up
04:43.02dijibwant pastebin to see further.
04:43.10dijiband no i havnt changed my naming convention yet
04:44.36ChannelZI think something is wrong with your macro.
04:44.51dijibno macro.
04:44.58dijibscope.
04:45.00ChannelZLike I said
04:45.01dijibhttp://pastebin.com/efm9vVGZ
04:45.19dijibthats error and extensions.conf. in whole
04:45.50ChannelZWhy do you post errors and then we find out it's something totally different?
04:46.02dijibany 877XXXXXXX or 519XXXXXX have been intentionally modified
04:46.28dijibthats not differernt its the same error\\
04:46.40WIMPyCould that be a failed re-invite?
04:47.12ChannelZlooks like
04:47.19dijibwhen a call comes inbound. it works for 30sec & then drops. i can call all internal and make outgoing without a problem
04:48.04kaldemarbesides the spawn extension is not an error.
04:48.04WIMPyThat again sounds like a rtptimeout, but it should say so.
04:48.40dijibthats verbose 47 or something
04:48.44dijibis that enough?
04:48.45snadgeis it normal for asterisk to just send off registration requests every second?
04:48.46ChannelZWhat's the extra ,60 on the end of your Dial for too
04:49.09dijib60 second timeout?
04:49.17ChannelZNot so much, no
04:49.29ChannelZThat's what the 20 is.
04:49.30dijibwait your right
04:49.33dijibafter options
04:50.04ChannelZSo you're saying there is 30 seconds of call time between lines 34 and 35?
04:50.05WIMPysnadge: You don;t seem to be able to communicate with the peer, or the peer doesn't want to,
04:50.42WIMPydijib: With full audio, in both directions?
04:50.59dijibyes.
04:51.06ChannelZIt never reports having answered the channel. Something's goofy
04:51.08dijibcall beyond the drops are okay
04:51.25ChannelZoh wait nevermind I am blind
04:51.32ChannelZcloses some windows
04:54.00ChannelZSIP debug?  Do they send you a BYE out of the blue?
04:54.06snadgeWIMPy: im really not surprised.. thats why this particular task requires an expert i think
04:55.05*** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net)
04:55.10dijibi can get logs of anything thats needed. although offsite
04:55.18ChannelZsnadge: does your net work?
04:57.03snadgeyes.. this same configuration was working previously.. i've had this particular problem before, but cannot remember how i solved it... or whether its just something that goes away by itself
04:58.07snadgehttp://whirlpool.net.au/wiki/iiNetPhone_asterisk
04:58.56snadgeiinet don't specifically support asterisk, and refer to this article.. and say "don't blame us if it doesn't work"
04:59.17snadgeit would be nice if someone with a brain could update the instructions
04:59.23snadgeyou know, so that they actually work ;)
05:00.24snadgei'd ring them up and ask them for help.. but they specifically say not to
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05:00.59snadgei guess they dont want people running asterisk exchanges off of their voip accounts
05:01.09snadgeand want them to use simple voip clients instead
05:02.18ChannelZHave you turned on SIP debug?  Do you get *any* response from them when you register?
05:04.41*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
05:07.23snadgei do occasionally
05:07.47snadgelike right now for example.. it says my outgoing trunk is "registered"
05:08.26snadgebut it keeps sending off requests anyway.. and now its in state "request sent"
05:08.29snadgei dont get it
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05:17.57ChannelZAre you behind a firewall?
05:18.02dijibme?
05:18.07ChannelZno snadge
05:18.10dijibk
05:18.58dijibwhats his issue? snadge ?
05:20.03ChannelZRandom registration errors, timeouts
05:21.40*** join/#asterisk oej (~olle@195.41.130.3)
05:22.00dijibbut can establish a connection to itsp at all?
05:22.08ChannelZsometimes
05:22.15dijibupnp?
05:22.22ChannelZsounds like network wonk to me.. his or theirs, dunno.  He's apparently napping now.
05:22.42dijiblet me have access to your router snadge
05:22.47dijibill open you up.
05:22.56ChannelZFilthy.
05:23.03dijibwhat?
05:24.07dijibmeaning udp 5060 && 10000-20000
05:24.42ChannelZHis reg sometimes works, qualifies sometimes doesn't.  I doubt it's "half closed" but something is going on
05:25.10dijibupnp would open them up then on reboot might be closed
05:25.30dijibget a wiresharkk>?
05:25.36ChannelZApparently his ITSP is not fond of Asterisk either so he's got that going for him.  If he'd come back I'd offer him to reg to me, rule out his ITSP
05:26.02ChannelZor rather I guess this is a VM or something, not an ITSP
05:26.05dijibso asterisk to asterisk. what do>?
05:26.24dijiboh is it an appliance?
05:26.33ChannelZno idea to be honest.
05:27.29dijibyouve never * <-> *
05:28.11ChannelZI go by what they say which in this case is not much
05:28.54dijibhave you ever setup two asterisk boxes able to call eachother internally?
05:30.01ChannelZFor fun yes
05:30.19*** join/#asterisk Juggie (~Juggie@unaffiliated/juggie)
05:30.29ChannelZOr.. well I guess practically too, I do have my home * connected to my work *
05:30.42dijibcool
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05:31.07dijibwhen you do a Dial how do you right it?
05:31.12dijibwrite
05:31.14ChannelZThough it still qualifies more for fun.
05:31.32ChannelZDial(SIP/myworkpeer/123) like anything else
05:32.17dijiband dialing rules? like 3XX,n,Dial
05:32.31ChannelZDepends on how you want to set it up.
05:32.54ChannelZFor instance at work I have a ** wildcard exten that just sends everything to my house
05:33.17dijiby?
05:33.18ChannelZSo I can dial **200 to dial exten 200 at home, or any exten
05:33.38dijibi would like to see your dialplan
05:34.17ChannelZexten => _**NXXNXXXXXX,1,Dial(SIP2/myhomepeer/${EXTEN:2})
05:34.44ChannelZfor dialing out my home number.  Similar exten on the 'home' side minus the **
05:35.31ChannelZ(well and it dials something different... SIP/myITSP/${EXTEN}
05:35.37dijibi think i need to find an asterisk for dummys
05:35.39ChannelZas appropriate
05:36.19*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
05:36.58ChannelZLike say you have SIP/ServerA and SIP/ServerB - all extens on ServerA are 1XX and all extens on ServerB are 2XX.  To dial an exten on ServerB FROM ServerA, you'd have an exten like _2XX,1,Dial(SIP/ServerB/${EXTEN}
05:37.55dijibwell ChannelZ thanks for the breif lesson. ill see if could incorporate this. nite
05:38.08ChannelZ${EXTEN} represents whatever extension was actually dialed (_2XX being a pattern) so dialing any 3 digit exten starting with a 2 just passes it along
05:38.35dijibahh l
05:38.37dijibk
05:38.45dijibi think i get it
05:39.17ChannelZJust a matter of setting up the proper peers on each side so you can route them into your dialplan correctly, and then Dial the peer with whatever exten
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05:51.41snadgesorry guys.. was on the phone to my gf
05:52.25snadgei am behind a firewall.. and the correct ports are open, and this configuration was previously working fine
05:53.55ChannelZso clarify what is what; you're running an Asterisk locally which is registering to an ITSP who hates Asterisk?
05:54.26WIMPyAre the ITSPs who don't?
05:54.26snadgecorrect
05:54.52snadgethis configuration was previously working too.. after messing around for a while, looking at various conflicting instructions etc.. then i get back from holidays, and its stopped working again
05:55.02ChannelZWell if you want to register to me and see if it works, it'll narrow down whether it's you or them.  I know my system works.
05:55.26snadgeok sure.. would you hate me if i said i was using freepbx
05:55.47ChannelZI hate the game, not the player :P
05:55.51snadgeastrisk@home actually ;) (hey.. shoot me.. i just wanted to get something up and running quickly)
05:56.17snadgei have set up other trunks before, which work fine though
05:56.19WIMPyThat's probably why ITSPs refuse to support Asterisk.
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06:19.48schmidtsgood morning
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06:25.17kleszczmorning
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06:40.15ChannelZaloha
06:40.31WIMPyCDMA-CD
06:41.14ChannelZYes
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06:53.45snadgeim totally going to reply to that iinet whirlpool thread by the way
06:54.03snadgeand say.. what crackpipe was the person smoking when they came up with that registration string
06:54.10snadgeand could they put the crackpipe down please
06:54.58snadgeand then give them a working registration string.. but i need to test the incoming calls first.. as soon as i get off the phone to queensland transport :/
06:55.05WIMPyA forum?
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07:14.42ChannelZHmm.  All the SIP probers seem to be from Hungary
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07:26.06snadgeyeah its a forum
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07:56.13olliitzafrir: hey .. i'm trying to use your dahdi-extra git repository for zaphfc and oslec modules...make gen-patch created a diff file which i want to use to patch generic dahdi-2.4.1 but it fails while patching Kbuild with some hunks...would you mind to pm me for more details?
07:57.27tzafrirollii, hmm... I've stopped mantaining it, and switched to https://gitorious.org/dahdi-extra/dahdi-linux-extra instead
07:57.45WIMPyYet another zaphft version?
07:57.55WIMPyI guess that makes it a no.
07:58.15tzafririt's the same code. But now organized as a clone of the dahdi code, which is really the right way to do it
08:00.38WIMPyIs what ollii just wrote te summary of that version?
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08:03.41WIMPywarning: remote HEAD refers to nonexistent ref, unable to checkout.
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08:06.56mutex7cHello everybody. Is there any best practice to limit simultaneous calls within the dialplan PER EXTENSION ? I see there are parameters to do this for SIP peers, which would allow global limitations for trunks etc. But I would rather be able to track concurrent channels in and out through specific extensions in the dialplan ... Any hints - The Google brought up nothing useful thus far .-o
08:07.42mutex7cOf course I could do some funky func_odbc ... But maybe any other idea ?
08:07.52olliitzafrir: okay .. good to know, i'll give it a try
08:08.40olliitzafrir: with dahdi-linux-extra: warning: remote HEAD refers to nonexistent ref, unable to checkout.
08:09.25tzafrirhmmm...
08:09.28tzafrirlet me see
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08:57.23kaiimutex7c: see functions GROUP() and GROUP_COUNT()
08:57.43kaiimutex7c: or just set call limits on your sip peers ... (not in the dialplan, though)
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09:30.38mutex7ckaii: thanks, I will look into that. I am now working directly with channel registrations to implement my own bandwith quota management regarding routet sip-ids ...
09:31.08mutex7ckaii: but the functions might come handy anyways - thanks for the hint :)
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10:17.17olliitzafrir: something new aber your extra repo? :)
10:17.44tzafrirlooking into that...
10:17.54olliigreat, thank you
10:20.12tzafrirgit checkout -b master origin/extra
10:20.58tzafrirthough the interesting thing is really:  git diff origin/svn_trunk origin/extra   #and such
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10:24.19WIMPytzafrir: What's the summary? Yet another zaphfc + oslec?
10:24.47tzafrirWIMPy, I want it to be a single repository for all "other" drivers
10:25.08WIMPyIs there a definition for "all"?
10:25.35WIMPySo is the zaphfc identical to another one?
10:26.47WIMPytzafrir: both master and origin/extra result in not found
10:29.01tzafrirATM: oslec, zaphfc, Aligera ap400 drivers, some OpenVox drivers, Voicetronix OpenPCI
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10:32.50coppicetzafrir: how long has aligera been around? I hadn't heard of them before
10:33.20WIMPyIt's easier to find tzafrir patch than the card itself.
10:34.18WIMPytzafrir: Does it include support for the OpenPRI as well?
10:36.43tzafrircoppice, frawd has been in touch with them.
10:37.21tzafrirMy rule is: someone has to be in touch with the vendor and be able to answer bug reports, update drivers, and such
10:37.23WIMPytzafrir, ollii: Just -b extra works.
10:38.13tzafrirWIMPy, OpenPRI: no DAHDI drivers for them, IIRC. But I guess VoiceTronix would be able to give you a better answer
10:38.55coppicetzafrir: it looks like they did what I tried to encourage several people to do - make simple card inside Brazil
10:38.57WIMPytzafrir: I just want to find out, what's in your repo there. But now that I got it, I'll take the README.
10:39.05tzafrir(One of the voicetronix guys is ron@debian.org and he's my point of contact, though he does not deal with the dahdi drivers)
10:39.41tzafrir(BBL food)
10:40.25WIMPyHmm. The README seems to be from the original dahdi.
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11:01.40tzafrirWIMPy, it's a clone (in the git sense) of dahdi
11:01.46tzafrirdahdi-linux, that is
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11:04.38WIMPytzafrir: I was just looking for information about what the additions are, as I think I dould list that under my download links. But I guess your first answer sums it up.
11:10.13tzafrirBTW: my main point of contact at openvox no longer works there. This is why the OpenVox drivers there are not exactly up to date
11:10.36tzafrirSo if anybody wants to pick that up, feel free to do so
11:17.10coppiceoh, those guys are just a few kilometres over there >>>>>
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11:41.45olliitzafrir: dont get it... :D 1st step: git clone after that git checkout -b master origin/extra ?
11:44.24WIMPyOnly -b extra
11:45.03olliiah okay..
11:45.11olliithanks, that works
11:46.47ollii# git checkout -b extra
11:46.47olliifatal: You are on a branch yet to be born
11:47.09olliiafter doing: git checkout -b master origin/extra # git checkout -b extra does it
11:57.33olliitzafrir: could this be consired as a stable branch?
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12:39.10aberriosWhat would be the preferred method of implementing "cli screen popping", listening to the AMI for the extension answering a call, some kind of output from the Dial plan or using the Queue URL param?
12:39.58aberriosinterface would be a browser...
12:40.13aberriosuser interface*
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12:53.18kaiiaberrios: i would prefer AMI, possibly utilizing one of those manager proxies if you plan to implement this for many users
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12:58.36makmak78hello! i have been troubleshooting my asterisk 1.4.36 2 weeks because my calls getting dropped randomly in middle of call. i have found that double INVITES is sent. help needed!
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13:03.39kaldemarmakmak78: double invites in the beginning of a call or are you seeing invites in the middle of a call?
13:04.25kaldemaraberrios: AMI events
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13:05.52makmak78kaldemar: in the beginning
13:06.24WIMPyFrom the same host?
13:07.17WIMPyAt the same time?
13:07.50makmak78Wimpy: yes
13:08.02kaldemarmakmak78: that's probably normal re-invite behavior if asterisk does not stay on the media path. enable sip debug and try to get a sip trace of what happens.
13:08.47kaldemaror maybe you should pastebin a sip debug of a call setup so someone can take a look at it.
13:08.57makmak78kaldemar: its these calls that gets hungup in midconversation
13:09.25makmak78i actually have a pcap file of these calls with all data
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13:14.13Kobazhow would i compile dahdi for a kernel other than the one i'm running
13:14.20Kobazis there an env variable for the kernel source path
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13:15.25kaldemarKobaz: set KVERS for make.
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13:16.55Kobazk
13:17.34kaldemaralso KSRC can be set. see the makefile, it's in the beginning.
13:17.39Kobazperfect
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13:18.58asilvaHello, can anyone tell me if there is any option on iax.conf that enables DTMF ? or it should work naturally ?
13:20.09Kobazdtmf should just work
13:26.27asilvaKobaz, can't even see them on logs, with sip works just fine.
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13:27.55makmak78kaldemar:it looks like this in short:
13:27.57makmak78>invite
13:27.57makmak78>invite
13:27.57makmak78<trying
13:27.57makmak78<rtp
13:27.57makmak78<session progress
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13:30.00WIMPy~pb
13:30.00infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
13:30.05WIMPymakmak78: ^^
13:30.41makmak78it thought it was a bit to short for pastebin
13:30.59makmak78apparently to long for mirc, anyway
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13:32.17[sr]dahdi 2.5
13:32.17[sr]:p
13:32.22[sr]update the topic!!! :)
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13:32.26russellbanonymouz666: what's up
13:32.32anonymouz666hello russellb
13:32.57anonymouz666using 1.8.6-rc2 in both machine A and B, using the distributed device state
13:33.10anonymouz666after injecting calls on machine A
13:33.58anonymouz666the command ais clm show members got stuck
13:34.03[sr]hi WIMPy, need you
13:34.21anonymouz666if I start ais "<tab>" and the CLI got stuck
13:34.37anonymouz666then i stop asterisk, restart aisexec and the things starting working again
13:34.46WIMPy[sr]: What's up?
13:35.13anonymouz666i am using the corosync 1.4.1 (the book is 1.2.8 if i remember) and asterisk 1.8.6rc1
13:35.30anonymouz666do you a suggestion on where I should start looking to figure out what's happening
13:35.43russellbanonymouz666: hm, don't know.  i'd say install latest corosync and latest openais, and if it still does that, file a bug on issues.asterisk.org
13:35.51russellbthough it seems unlikely that it'll get looked at soon ...
13:36.53anonymouz666it is already in the latest version of both
13:37.10anonymouz666but even to file bug is hard, because there's no core dump
13:37.15anonymouz666things got stuck
13:37.18anonymouz666until a restart is done
13:37.37anonymouz666can I follow the general deadlock instructions ?
13:37.58russellbyou can try, it could be a deadlock
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13:38.07russellbi would include a "thread apply all bt" from gdb
13:38.31[sr]WIMPy: remember i ask about havind the convencional PBX and an ISDN phone connected to the NTBA?, ok works great, but in my base, i want to catch a number, that jumps from one to other NT, in my case i have two NTBA's
13:38.50[sr]WIMPy: asked the ISP and there's no way they can configure it so that number always hit a certain NTBA
13:39.14beekHey--- russellb is slumming it today!
13:39.16[sr]for my backup system, there's 50% chances that i lose a call, in a backup scenario
13:39.17WIMPy[sr]: With an extra PBX? I thought only a phone?
13:39.18anonymouz666russellb: alright, thanks for the information.
13:39.32russellbbeek: hm?
13:39.54beekBack in the neighborhood...
13:40.10beekWill we see you as Astricon this year?
13:40.11[sr]WIMPy: hum a solution is a phone with two ISDN line in connections
13:40.18singlerhow could I start second asterisk on same server? Should I try chrooting it? I copied directories, modified asterisk.conf and try to run "asterisk -C /etc/asterisk2.conf", it runs, but it creates lock in /var/run/asterisk instead of /var/run/asterisk2, so normal asterisk process cannot start..
13:40.19WIMPy[sr]: If you cut the line in front of the other NT, i.e. the line from the telco, they shouldn;t send calls there any more.
13:40.33WIMPy[sr]: I don't think that exists.
13:40.51Kobazthe con
13:40.52[sr]WIMPy: i'll lose 4 call's at a time with that
13:40.58Kobazastri of the con
13:41.16WIMPy[sr]: Pardon?
13:41.34WIMPysingler: That will certainly work.
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13:42.18[sr]WIMPy: i could disconnect the 2nd NT, and that would work, already tested, but i wouldn't have 4 simultaneus call anymore, only 2
13:43.02WIMPy[sr]: Yes, but how many phones do you connect?
13:43.31singlerWIMPy: do you mean that it will certainly work with croot? Is there a way to start second asterisk without chroot?
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13:44.36WIMPysingler: I'm not sure if you can twek the configuration enough to do without. But with chroot it has to work.
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13:45.26singlerok, thnx. Weird thing is that it does not use configured /var/run directory, guess I will need to setup chroot
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13:48.25treborsuxI installed trixbox ce I ahave an openvox ae81p  How do I get the drivers working?  I thought the drivers were included already
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13:49.10jayteetreborsux, try asking in
13:49.18jaytee#trixbox
13:49.26treborsux15 people all sielent
13:49.43treborsuxanyone know what distro does include drivers for that card
13:49.44jayteewell, almost no one here uses trixbox
13:49.53beekmornin' jaytee
13:49.57jayteemornin beek
13:50.17jayteetreborsux, not sure. did you look on openvox's website?
13:51.50WIMPytreborsux: git clone git://gitorious.org/dahdi-extra/dahdi-linux-extra.git -b extra dahdi-linux-extra
13:52.05treborsuxya It confuses me says it is for tribox and certified by tribox but then only manual has instructions for recompile
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13:52.09WIMPyUnless it is a standard hfc card.
13:52.42jayteetreborsux, there is a pdf document on the openvox website for installing the a800 device driver on trixbox.
13:52.53WIMPyCan't find that card.
13:52.59[sr]WIMPy: the NTBA's have 2 connectors right? my ideia is to have one of them of each NTBA connected to a PBX, asterisk or not, and the other connector to the ISDN phone
13:53.03[sr]got the idea?
13:53.15treborsuxthere is but like if i installed linux not if i installed ce
13:53.24WIMPy[sr]: Yes, ok.
13:54.22[sr]but have the problem of the number jumping from one NTBA to the other
13:54.41[sr]two ISDN's phones solves the problem..
13:54.53treborsuxit's an a810p card btw
13:54.57WIMPy[sr]: Yes, hence my suggention to cut the other NT in case te PBX is down.
13:55.35[sr]WIMPy: ah, whem it's down, ok it's a solution, didn't read the "when down" before
13:55.35[sr]sorry
13:56.07WIMPy[sr]: Well, I thought you got that idea yourself :-)
13:56.48[sr]i was thinking you said to cur the 2nd NT forever, thats why i was saying about the 4 call cimultaneous
13:56.53[sr]cur=cut
13:57.38WIMPyThat doesn't make sense.
13:57.43WIMPy:-)
13:58.08treborsuxbash no command get
13:58.18treborsuxgit i mean
13:58.28[sr]WIMPy: i'l still sleeping!! pardon me
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13:59.12WIMPytreborsux: Then install git. But I have a feeling that won't be the only thing missing to get the stuff installed.
13:59.36[sr]WIMPy: a beatiful solution was a primary access, but it's expensive :S in here the ISP only sells primary access's with half of the channels activated, thats the mininmum
13:59.45treborsuxwhats the word on asteriskwin32?
14:00.01kaldemartreborsux: forget about it.
14:00.06treborsuxok
14:00.31treborsuxhmm looking for a ditro that supports that card
14:00.36WIMPy[sr]: I don't think you can get fractional PRIs here.
14:00.52treborsuxI am not very good with linux.  Windows network admin
14:00.56[sr]WIMPy: they only sell the full 30/32 channels there?
14:00.57WIMPytreborsux: I wouldn't have much hope.
14:01.09treborsuxI find myself just following instructions not know what it is doing
14:01.10WIMPyyes
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14:01.38WIMPytreborsux: Then you probably got the wrong card.
14:01.46[sr]i see, here they do, but with the minimum of half of it
14:01.55WIMPy[sr]: If you only need half of it, 8 BRIs would be cheaper.
14:02.00treborsuxOpenvox AE810P
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14:02.14[sr]WIMPy: well, in here they don't self more then 3 BRI's
14:02.17[sr]stupid i know
14:02.19treborsuxI guess ill have to go from scratch
14:02.34[sr]if someone needs more then 3BRI's, they upgrade to PRI
14:02.42[sr]i think the price is +- equal
14:02.54[sr]3 BRI or 1x PRI (with half channels activated)
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14:03.05treborsuxI was hoping for it being in a turnkey distro
14:03.17WIMPy[sr]: Are the BRIs so expensive or the (half) PRIs so cheap?
14:03.24treborsuxANyone get crazy and put asterisk on ubuntu
14:03.54WIMPytreborsux: You can run Asterisk on any *X.
14:04.01treborsuxI know
14:04.20[sr]WIMPy: no idea which are, but one BRI cost's about 30/32€month+VAT, 1 PRI with half channels about 80/90€/month+VAT
14:04.21WIMPytreborsux: But if you want to use PCI cards, cut that to any Linux.
14:04.31[sr]this was the last price when i asked
14:04.32lanmowerI'm trying to connect my linphone to my remote *, I have forwarding set up to my linphone pc on 5000:5100 10000:20000 and 3478:3479 for good measure, my call drops after a few seconds. any ideas?
14:05.19WIMPy[sr]: Sounds cheap. The standard price for a PRI here is 300. But I got them offered for 90 if I take 5.
14:05.24*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:05.24*** mode/#asterisk [+o leifmadsen] by ChanServ
14:05.25lanmowerif it would help to mention my ping between the two points are relatively high (over 150ms and under 600ms)
14:06.04lanmowerI'm also using dyndns on both ends.
14:06.49*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
14:07.35lanmowermy linphone is configured for 5060, 100008, my asterisk is running on its default port and confirmed working with a service providers trunk and local extensions.
14:08.00treborsuxElastix?
14:09.03WIMPytreborsux: I recommend you change the card for another brand or get a Linux guy to get it working.
14:09.14lanmowerit seems strange for a call to run for a while and then stop, sip debug just shows bye's, calls last along the lines of 10 seconds then die.
14:09.32[sr]WIMPy: come live in here :p
14:10.07WIMPyWhat?
14:14.00WIMPy[sr]: Where should I live? On IRC or in yor company?
14:14.27p3nguinIn his computer case, of course.
14:15.22WIMPyIf it's the case for a Zuse or something, that could work.
14:15.44*** join/#asterisk Godfather_ (~estanteri@90.170.34.92)
14:15.59p3nguinas400 case?
14:16.27WIMPyNope
14:16.47p3nguinBigger?
14:16.56WIMPyyes
14:18.22*** join/#asterisk logicwrath (~no@c-68-62-24-205.hsd1.mi.comcast.net)
14:19.41[sr]WIMPy: hum choose :p
14:20.13p3nguinlike a z3?
14:20.51WIMPyThat might be ok.
14:22.04p3nguinThe Z3 pictures I see look like the cases aren't very deep.  I think you'd have more room in an old AS/400.
14:23.15WIMPyThe Z3 already had a case?
14:23.49p3nguinMaybe these cabinets are aftermarket?  I don't know.  The pics I'm finding have plastic cabinets.
14:24.05WIMPyIndeed, that's too modern.
14:24.11p3nguinOh
14:24.45p3nguinYou want to live in a computer with an entry door where you can walk inside and change out tubes.
14:25.00WIMPyThat's more like it.
14:25.09*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
14:25.19p3nguinYou may have to pay rent.
14:26.48treborsuxI just hate typing
14:26.54treborsuxIll make it work
14:27.13treborsuxI was just hoping for a little more turnkey action
14:28.17p3nguinDid you check if it is supported in AsteriskNOW?
14:28.52treborsuxI installed that it isnt turnkey
14:29.07treborsuxbut I couldnt find instructions to install it
14:29.13Kattyhello
14:29.19Kattymy asterisk does not work at all, how to fix plz???
14:29.21treborsuxonly instructions if i install aterix ground up
14:29.33beekKatty: Hire someone to fix it for you!
14:29.35beek:D
14:29.41WIMPyhands Katty a biiiig hammer
14:29.41Kattywhat is hire???
14:29.59treborsuxI can type for 4 hours and make it work just trying to avoid it
14:30.02treborsuxjust lazy
14:30.03Kattywoot! hammer
14:30.22WIMPytreborsux: For how long have you been typing here?
14:30.24KattyHAMMER TIME
14:30.31treborsuxgood point
14:30.38[sr]gives his 10kg hammer to WIMPy
14:30.41treborsuxi am just getting encouragement
14:30.44treborsux:>
14:31.02Kattywatches WIMPy wish it was 10k
14:31.26Kattywhile we're on the topic
14:31.30Kattyall you boys that have ladies.
14:31.36Kattyget her something shiny for christmas (=
14:31.38*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
14:31.46*** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave)
14:31.50Kattytake your tail to jcpenny, and get her something PRESTTY
14:31.52KattyPRETTY
14:31.55WIMPyKatty: I have already used such a thing to remove a wall that stood in my way.
14:32.01MaliutaKatty: I'm getting something special for my ex ... a restraining order
14:32.04beekHow about a polished aluminum beer can?
14:32.14KattyMaliuta: that is acceptable.
14:32.29KattyMaliuta: make sure you have a celebratory grill out when it's all done
14:32.36Kattybeek: unacceptable.
14:32.40MaliutaKatty: she's already under a temporary, just going to make it permanent to go that extra little bit
14:32.59Kattyshe must be one hellevawoman!
14:33.11Maliutas/woman/psycho/
14:33.14Maliuta:)
14:33.25Katty*hee*
14:33.43Kattyone of these christmases i am going to get something shiny
14:33.51Maliutathinks Kattys surname may be "Jackson"
14:33.55Kattyand EVENTUALLY someone besides my mother and best friend is going to get me flowers.
14:34.18Maliutayeah, why do no guys who aren't me buy flowers anymore?
14:34.43Kattyi don't know
14:34.51WIMPyBuy? I prefer to grow them myself.
14:35.04MaliutaI've been told on more than one occasion "nobody has ever bought me flowers before"
14:35.10beekWIMPy: Those flowers are for display, not to be smoked.
14:35.12WIMPyThe trouble is that I don't find enough takers.
14:35.34MaliutaWIMPy: my idea of gardening his high concentration defoliant ... and an axe
14:35.51WIMPybeek: Nothing for smoking here. You could try the Datura if you dare.
14:35.51Kattynot too good at gardening either :<
14:36.45Kattywhat's the equivilent of shiny and flowers, for males?
14:37.01MaliutaIf everyone takes a look at a map of Australia for me ... you'll notice the effect of my main attempt at gardening. It's that big deserty part in the middle and towards the west.
14:37.01treborsuxa bj
14:37.15treborsuxthat is all we really want
14:37.17MaliutaKatty: geek toys
14:37.21p3nguina new distributor or carburetor
14:37.29MaliutaKatty: they're shiny
14:37.38Kattymakes mental note
14:37.45treborsuxok everyone lie
14:37.53p3nguina bj and a new carburetor would be even better!
14:37.57MaliutaKatty: at the moment NERF is high on my list
14:38.00treborsuxi heard that
14:38.10Kattyooh nerf. never thought of that
14:38.13Kattytakes notes
14:38.38MaliutaKatty: and the little usb rocket launchers work well
14:38.50MaliutaKatty: botique beers
14:39.34beekI want the Nikon 11-23mm F2.8 lens
14:39.38p3nguinA nice local beer is always welcome.
14:39.41beeks/23/24/
14:40.06Maliutabeek: no, you want a 1000mm F1 lens
14:40.12WIMPyOooh, that sounds expensive.
14:40.29*** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn)
14:40.34MaliutaF stop 1 photo's are mad
14:40.35beekMaliuta: I'd be happy with the 11-24 but if a 1000mm F1 comes in, that would be great.
14:40.37Kattyso far my plans are inflatible kayak, blues jersey, epic meal time sauce boss shirt
14:40.41Kattyand something off thinkgeek
14:40.59Maliutabeek: what are you shooting that needs a macro lens?
14:41.03Kattya case of seasonal beer would be a nice touch
14:41.15p3nguinI don't like seasonal beers.
14:41.15Maliutabeek: personally I've never had need for anything below 28mm
14:41.25WIMPyOh, there has been a F0.8 Lens.
14:41.34Kattyi'm doing little Gift Bags for people this year.
14:41.36p3nguinBut a year-round beer from a local brewery would be good.
14:41.45MaliutaKatty: what's in mine? ;)
14:42.01Kattyi had planned chocolate oranges for everyone
14:42.02eduzimrsanyone here has * running with sip realtime in cluster (active-passive) with rsync?
14:42.04Kattyand something off thinkgeek
14:42.15Kattythen something from bbw for the ladies
14:42.23Kattywasn't sure about the guys tho
14:44.29beekMaliuta: Occasionally I need a really wide lens.  I've rented the 11-24 and really like it.
14:44.55WIMPyThese super wide ones are fun.
14:45.04beekYep.  And that one sings.
14:45.29Kattywhat's something that just about any guy would like, in the 10-15 price range?
14:45.43beekBeer
14:45.44_Corey_booze
14:45.57beek^5 _Corey_
14:46.08_Corey_:)
14:46.18Kattyso what, just wrap a case of beer, and put the gift bag on top?
14:46.21beekSee Katty -- guys have simple tastes.
14:46.23beekThat's be fine.
14:46.37_Corey_you could put a bow on it but it will probably go unnoticed
14:46.49Kattyi should order special wrapping paper from thinkgeek
14:46.54Kattythat might get noticed
14:47.09Kattyi wonder what schnucks would think with me walking out with 10 cases of beer
14:47.18Kattycause you can buy the big cubes for 15 bucks, right?
14:47.39_Corey_uh, well...  if you're hosting a game of "beer pong" maybe
14:47.58_Corey_if I'm getting beer as a gift, I'd want something a little more enjoyable :)
14:47.58KobazKatty: 10-15 hmm.  I don't usually buy stuff other than food
14:48.32_Corey_I'm getting old though, so it's definitely an age thing...  Anyone under 23 will probably be happy with a $15 cube of beer
14:48.40p3nguinMy concern is that you don't have a refrigerator large enough for 10 cases of beer.
14:48.45Kattynone of my friends are under the age of 23
14:48.56Kattymost of them are in the 28-35ish range
14:49.11KobazI think the lumberjacks are here
14:51.08Kattymaybe stick a nerf gun in everyone's bag!
14:59.53Qwellwtf is a cube of beer
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15:00.51WIMPywonders if they still have those "keggies" here.
15:01.35p3nguinqwell: 30-pack
15:02.15Qwellbeers don't come in "packs" higher than 6.  Don't lie.
15:03.03*** join/#asterisk datalay (~datalay@unaffiliated/datalay)
15:04.49datalaywhat is the cheapest way to connect to PSTN with Elastix,  i use cisco 3102SPA but it s 100$ :(((((((
15:05.00datalayor with Asterisk
15:05.18Qwellan ATA, or get an ITSP
15:05.42p3nguinThe SPA-3102 is a reasonable piece of hardware for that purpose.  I found mine for $70 US.
15:06.14WIMPyThe cheapest is a BRI card if you live in the right area.
15:06.37p3nguinIf you don't need to have a phone line for something special, you can go all VoIP and let someone else worry with the connection to the PSTN.
15:06.55[sr]brb
15:07.03Qwell[sr]: hurry back!
15:07.11[sr]i'll :p
15:09.34treborsuxwhat is no hardware timing source found in
15:10.50p3nguinIs there more to that sentence?
15:11.21WIMPy/YOU/ call that a sentence???
15:11.34*** part/#asterisk kdmessano (~nonya@unaffiliated/kdmessano)
15:11.48p3nguinIf all the words were presented, I might.
15:13.38*** join/#asterisk nullslash (~nullslash@pdpc/supporter/student/nullslash)
15:15.31nullslashHello, I'm looking for any cheap SIP provider. Could you suggest one for me?
15:16.11WIMPy~itsp
15:16.12infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
15:16.12timholumnullslash: voip.ms is inexpencisve, but I recently switched to nextvortex for the better quality
15:16.19WIMPynullslash: ^^
15:17.40treborsuxwhat is no hardware timing source found in //proc/dahdi loading dahdi dummy
15:18.10p3nguinSounds to me like you didn't install dahdi and/or load the module.
15:18.28WIMPytreborsux: You don;t have any hardware running that provides a timing source.
15:18.38treborsuxit is asterisknow isnt it already installed?
15:18.51treborsuxdo i need to
15:19.01treborsuxi dont need to hook a clk to this card do i?
15:19.24p3nguinIf there were punctuation in the "sentence," perhaps I would have come up with the same interpretation that wimpy did.
15:19.29nullslashtimholum, Thanks, but I will use it for residential phone.
15:19.55nullslashtimholum, what's wrong with voip.ms?
15:20.09WIMPytreborsux: No, you don't connect enything external, usually. But I can't comment on your card.
15:20.33p3nguinIt could have made more sense if it said, "No hardware timing source found; loading dahdi dummy."
15:20.42timholumFor a home user, nothing. It just would drop a call or two every once and a while ( every 500 phone calls or so )
15:20.46treborsuxso that doesnt matter than that it says that
15:20.58timholumI still use them for my failover
15:21.11timholumthey are 1.5c / min
15:21.51timholumhttp://www.voip.ms/
15:22.55timholumohh, and it looks like they droped there price, they are 1.05c / min
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16:25.08SuperNullsoooo ... if i wanna do caller id blocking but still be able to bill the call with the correct source # .. how would i do this ;-)
16:31.31*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
16:32.28WIMPyWhere do you want to block it?
16:32.45WIMPyAnd from an to where do you want to call?
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16:57.14treborsuxdamnit there is no txt editor in elastix distro and i cant seem to install one too many dependencies
16:57.32Qwellwhat
16:57.35treborsuxis there one already there i couldnt install pico
16:57.47treborsuxi need to edit a file
16:57.58Qwellnano, vi/vim
16:58.09treborsuxi installed elastix
16:58.16treborsuxi need to edit modules file
16:58.19_Corey_I haven't seen pico in a distribution in like 10 years
16:58.36coppiceis vi a text editor? I thought it was a medical treatment for high spirits
16:58.36treborsuxsorry that was the last time i installed linux
17:01.12_Corey_there was some licensing spat with Univ. of Washington who owned pine/pico, so nano replaced it
17:02.27jayteetries to think of a current linux distro that doesn't have nano.......fails.....
17:02.50Qwellmost embedded stuff ships with vi
17:02.54Qwellbusybox vi
17:03.25_Corey_yeah, a lot of the recovery distros and other slim installs lack nano so I usually tell people to download a VI reference card and suck it up :)
17:08.10*** part/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl)
17:15.49anonymouz666joe asterisk.txt
17:16.14anonymouz666edit autoexec.bat
17:16.35p3nguinI would be surprised it it didn't have joe, ed, or ee.
17:17.43WIMPyWhere are the emacs guys?
17:18.51QwellWIMPy: Why would emacs people get into an editor war?
17:19.12WIMPyGood point
17:20.31WIMPyBut then, I was pretty astounded, when I found out that elvis does "wysiwyg" HTML editing a few years ago.
17:21.38p3nguinCan someone explain to me why asterisk can no longer run mutt from System()?  I found out that it can run sudo -u asterisk mutt from System(), though.  I'd really like to know why, and what changed that makes it not be able to run it directly anymore.
17:22.05Qwellsudo creates a proper shell
17:22.14Qwellrather, shell env
17:22.40p3nguinbut asterisk's shell is /bin/false, so I don't understand that, either.
17:22.58Qwellasterisk users shell != shell Asterisk uses
17:24.01p3nguinI'm certain System() had no problem running mutt in the past.  Was there some type of security fix that stopped it during the past 1-2 years?
17:24.17Qwellno, but mutt may have done something that made it require a better shell env
17:24.23p3nguinoh
17:24.27p3nguinI hadn't thought of that.
17:24.58p3nguinRunning !mutt from asterisk CLI works, so I was really at a loss for explanation.
17:25.23QwellSystem and ! do things differently
17:25.46p3nguinI don't know asterisk that intimately, so I didn't know that.
17:28.56treborsuxi type dmesg and i can only see last few lines
17:29.03Qwelldmesg |less
17:29.04treborsuxhow do i see this page by page??
17:29.04p3nguindmesg|more
17:29.12QwellIt's 2011.  Nobody uses more.
17:29.19p3nguindmesg|most
17:29.25Qwelldmesg|some
17:29.45p3nguinerror: package 'some' was not found
17:30.00p3nguinNot available.
17:30.14*** join/#asterisk xnfinite (~xnfinite@41.29.223.87.dynamic.jazztel.es)
17:30.44Qwellfor i in $(seq 1 $(dmesg | wc -l) 10); do dmesg | tail -n$i | tail -n-10; done)
17:30.51Qwellminus the last )
17:31.37p3nguinWhen asterisk is configured to run as its own user/group, does it still start up as root first, then drop privs?
17:32.47Qwellfor i in $(seq 10 10 $(dmesg | wc -l)); do echo $i; sleep 1s; dmesg | head -n$i | tail -n-10; done
17:32.50QwellAlso, that is the winner.
17:32.56Qwellminus the echo/sleep
17:33.03Qwellstupid debugging code
17:33.12Qwellfor i in $(seq 10 10 $(dmesg | wc -l)); do dmesg | head -n$i | tail -n-10; done
17:33.13QwellTHERE
17:33.19Qwelldmesg, page by page.  10 lines per page.
17:33.31Qwell</self_amusement>
17:33.48jayteecopies another of Qwell's gems
17:44.26*** join/#asterisk jkroon (~jkroon@dsl-241-229-106.telkomadsl.co.za)
17:46.53eduzimrsanyone here has * running with sip realtime in cluster (active-passive) with rsync?
17:48.22anonymouz666the chapter 22 in the TFOT 3rd edition should be expanded and become a book about the subject.
17:53.09acidfoowhy the heck res_config_sqlite isn't in the 'resource' menu of make menuconfig ?
17:54.01Qwellacidfoo: I do believe it's in addons.
17:54.15Qwell~book
17:54.15infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook
17:54.34Qwellanonymouz666: ahh, clustering.  yeah
17:55.11acidfooQwell, addons is another asterisk package ?
17:55.23wizbitp3nguin: back from work
17:55.29acidfoook, well got it
17:55.33Qwellacidfoo: before 1.8 it is
17:55.37acidfoook
17:55.50acidfooah!
17:55.53acidfoothere we are
17:55.57wizbitConnected to Asterisk 1.8.5.0 currently running
17:55.58wizbit:D
17:56.21wizbitp3nguin: all the configs from 1.6 work
17:56.23acidfooQwell, and I guess that the addons version match the asterisk version? 1.4.3 is for asterisk 1.4 ....
17:56.30acidfooand 1.6.2.3 is for asterisk 1.6
17:58.44treborsuxin dahdi-channel.conf if i have 8 fxo ports i need to change all that to fxo but i see a note that fxs ports use fxo signaling.  If I use fxo ports it still stays fxo signaling?
17:59.19*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
18:00.37jayteefxo ports use fxs signalling and vice versa
18:00.44treborsuxok kewl
18:00.53treborsuxso i need to set signalling to fxs
18:01.10jayteeif it's an fxo port then yes
18:05.06anonymouz666Qwell: you will talk about cororsync and openais?
18:05.20anonymouz666*corosync
18:05.23Qwellyes
18:05.26acidfooerm
18:05.28acidfooCredits
18:05.28acidfoores_config_sqlite was developed by Richard Braun at the Proformatique company.
18:05.36acidfooI should ask insternally, im working there ;P
18:09.33anonymouz666Qwell: I am using both also. Having some problems in a stress testing... but things will get better, I think :)
18:18.11wizbitim sure the command 'reload' worked in 1.6
18:18.19treborsuxall looks good
18:18.21treborsuxbut
18:18.29treborsuxecho set to oslec
18:18.36treborsuxand it says failed
18:18.56treborsuxthis card has the echo hardware on it
18:18.56treborsuxwhat do i need to set this to?
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18:19.16acidfooin what application the function SetIfEmpty() is ? thank you.
18:19.32acidfoo(asterisk 1.4.42)
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18:19.47treborsuxOctasic HWEC  is what it has
18:20.19treborsuxso what do i set echo can to?
18:21.20treborsuxwhat do i set genconf_paremeters to for echo?
18:21.20Katty!
18:21.23Kattyhai
18:21.28Kattythis is not the console window
18:21.53treborsuxdriver loaded and i see it in elastic I am so proud of myself!
18:27.00p3nguinYou're using Elastix... you no longer have the right to be proud of yourself.
18:27.21p3nguinacidfoo: Would you rephrase the question so I can make sense of it?
18:28.06acidfoop3nguin, I realised it was an addon developped internally... don't worry much about it ;/
18:28.08treborsuxwhat do i set genconf_paremeters to for echo?
18:32.20treborsuxwhen set to oslec it says no
18:32.30treborsuxgot ripped on this card?
18:33.38treborsuxor is there somewhere else i need to turn on oslec also?
18:33.45treborsuxsome other setting in dahdi?
18:34.02treborsuxhttp://www.ebay.com/itm/A810P-8-Port-FXO-FXS-W-Octasic-HWEC-Card-Asterisk-/180686009944?pt=LH_DefaultDomain_0&hash=item2a11b9b658
18:34.17treborsuxthats what i bought i can see the oslec module on it
18:36.28QwellTry buying real hardware next time.
18:36.43Qwell•3-Month “No Question Asked” Return Policy
18:36.56QwellTake advantage of that.  Ask again when you have hardware that works.
18:37.06*** join/#asterisk guax (~guax@unaffiliated/guaxinim)
18:37.52Qwell(When you buy clone hardware, we get to laugh at you when it fails.  That's how this works.)
18:37.55guaxI have a question about transcoding cards. Does anyone know if i can scale up to 200 simultaneous calls (400 channels) with recording in one machine using them?
18:38.18Qwellguax: There is no coded limit that would prevent you from doing so.  Feel free to try it.
18:40.39treborsuxso openvox is imaginary company?
18:40.47QwellNo, it's a real company.
18:41.00treborsuxIs oslec something I have to install
18:41.13treborsuxwill it undo the compile of dahdi i already did
18:41.35treborsuxis oslec installed when i install dahdi
18:41.44treborsuxnothing has failed yet
18:41.57treborsuxi just dont think i installed oslec
18:42.55treborsuxwait i am confused if i have hardware cancel I dont use oslec
18:43.03jayteeoslec is software based echo cancellation, Octasic is a hardware echo cancellation chipset used by many vendors for hardware based echo cancel
18:43.07treborsuxlol
18:43.10treborsuxi am an idiot
18:43.26treborsuxso i set it to none
18:43.38treborsuxbecause i have hardware echo
18:43.43treborsuxRight
18:43.48jayteethat's what I did for Digium T1 cards with hardware echo cancel
18:44.12jayteeand they worked like a champ and AFAIK they still do years later
18:48.19radenKatty, :D :D :D : D
18:49.39*** part/#asterisk xnfinite (~xnfinite@41.29.223.87.dynamic.jazztel.es)
18:51.46Kobazanyone here use ipmi?
18:54.16jayteeIntelligent Platform Management Interface? or International Precious Metals Institute?
18:55.19p3nguinImitation Pizza Makers, Incorporated
18:55.51justdaveis there a way in Meetme (programmatically on the back end or otherwise) to mute everyone in a conference room in such a way that allows them to unmute themselves afterwards?
18:55.56*** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com)
18:56.08justdavetelling meetme to mute them puts an administrative lock on it so they can't unmute
18:56.28*** part/#asterisk guax (~guax@unaffiliated/guaxinim)
18:56.36justdaveand having people default to muted when they enter the room only works until someone unmutes to talk and then forgets that they're unmuted.
18:57.26justdavetrying to repair the problem by social reinforcement hasn't been working, so people are asking me for way to do it for them
18:57.45justdaveapparently we have people who join conferences and leave their phone connected while they walk away to answer the door and that sort of thing
18:57.47Kobazjaytee: the platform stuff
18:58.18jayteeKobaz, I haven't used it but Intel has a ton of stuff about it on their site
18:58.23Kobazyeah
18:58.26Kobazi can't get it to like, turn on
18:58.38Kobazin the bios it's got an ip address but nothing responds on it
19:00.27*** join/#asterisk espro (~hyrax@cpanel.vmlinuz.co.uk)
19:01.37esprois there someone i can chat with about t38 faxing?
19:02.08esprodifficult to find current documentation on how to get this going properly, so much of it is out of date
19:02.19*** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca)
19:03.52justdaveespro: as far as I know, the only stuff that actually works has to be purchased, and the vendors usually have good docs they ship with it.  I haven't tried it personally, but the rumors I keep hearing are that the freely-available stuff doesn't work very well. (and the commercial stuff doesn't always, either)
19:05.45esprojustdave: that's what I've been suspecting. I was looking at another commercial offering (hylafax enterprise), and while it works, they're asking for mandatory maintenance/support contracts which brings it to about $2000. decided to take a crack at the open source stuff again
19:06.15esproi saw digium's res_fax_digium, but i can't even get the damn thing to detect the license file at this point
19:06.33p3nguinWhat have you done to get it to use the license?
19:06.55p3nguinAlso, what asterisk version are you using?
19:07.37esprofollowed the docs to a tee pretty much, got a free license, downloaded the register app. it created the license in /var/lib/asterisk/licenses
19:07.40espro1.8.4ish
19:08.13p3nguinSo you built res_fax into asterisk when you compiled it?
19:08.32esproRPM off EPEL, so I have asterisk and asterisk-fax packages. res_fax.so exists
19:08.57espromessage I get in the logs is, "res_fax_digium.c: Failed to initialize res_fax_digium copy protection!"
19:09.05p3nguinhmm
19:09.15esprogoogle search led me to believe that has to do with the license, but I could be off
19:09.32p3nguinIn the asterisk CLI, run "module show like fax" just to see what is actually loaded.
19:10.06esprores_fax res_fax_digium res_fax_spandsp
19:11.02p3nguinI could be mistaken, but I think res_fax_digium and res_fax_spandsp will conflict.  Unload the spandsp one, and add a noload for it in modules.conf.
19:11.36p3nguinAfter that, unload res_fax_digium, them unload res_fax.
19:11.56esprohmm think the docs said it was app_fax that conflicts, i'll give it a shot though
19:12.12p3nguinapp_fax conflicts with res_fax.
19:12.49p3nguinI'm interested in seeing what they say as they load.  So once unloaded, module load res_fax.so, followed by module load res_fax_digium.so.
19:13.36p3nguinIf you can't get ffa to work, and if you want to go an alternate route, lose res_fax_digium and use res_fax_spandsp instead.
19:15.32p3nguinLots of people like res_fax_spandsp, so it shouldn't be too hard to find info on using it.
19:15.44esprop3nguin: still the same error
19:16.24*** join/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu)
19:16.27p3nguinWill you pastebin everything from where you load res_fax all the way to the end of what loading res_fax_digium spews out?
19:16.38esproyep doing that already
19:16.57esprohttp://pastebin.com/PSLZtWiF
19:17.58p3nguinLimiting to 0 sessions!  That's a problem.
19:18.12esproyeah, must be because it can't figure out the license
19:18.37p3nguinWhen you ran the registration thing, did it say anything that indicated a problem?
19:18.58espronope, everything looked to be okay
19:19.16esprofunny, think i solved it
19:19.41esprothat post i mentioned said that it expected licenses in /usr/share/asterisk/licenses, all the digium docs say /var/lib/asterisk/licenses, and that's where register puts it
19:19.54esprojust symlinked the /usr/share dir to it and it seems to be happy
19:20.32p3nguinMine is in /var/lib/asterisk/licenses/
19:20.48esproyou use the digium module?
19:20.58p3nguinI use res_fax_digium.so, yes.
19:21.04esproon what distro?
19:21.34p3nguinAlthough that's completely irrelevant... it's Arch Linux.
19:21.50esprowell, did you build from source or use a package?
19:22.06p3nguinIt's binary, as far as I know.
19:22.12p3nguinBut I made it into a package to install it.
19:22.25esproyou astvarlibdir setting in asterisk.conf is set to /var/lib/asterisk then i guess?
19:22.37p3nguinyes
19:22.39p3nguinas it should be
19:22.45esproseems the EPEL rpm points that to /usr/share/asterisk
19:22.48esprowhich is the problem then
19:23.21*** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn)
19:24.09p3nguinIs all the stuff in that directory rather than the appropriate place?
19:24.32esproi'm new to asterisk, but if you're referring to agi-bin, firmware, keys, etc, yes it's all in /usr/share
19:24.36espro/var/lib/asterisk is empty
19:24.44esproexcept for licenses (created by `register`)
19:24.51p3nguincrazy
19:25.10ChannelZcore show settings
19:25.15p3nguinI'm curious whose idea that was, and why.
19:25.19ChannelZ'data dir' probably?
19:25.47esproChannelZ: varlibdir, datadir, agidir all point to usr/share
19:25.57ChannelZso there you go
19:26.11esprostandard place however is var/lib ?
19:26.24ChannelZyes but apparently your asterisk was built differently
19:26.38ChannelZI missed most of the conversation, was it a package or from source or..
19:26.57esproChannelZ: RPM from EPEL
19:27.16ChannelZah.  Well they moved it :)
19:27.31esprousr/share isn't a logical place for it to be in my opinion so i'll just move it to the standard spot
19:27.37p3nguinWhile I do not condone installation from source, I do recommend building from source.  Build it the way you need it, then package it for your distro, then install the package.
19:28.56ChannelZscrew that, too much work
19:29.14esproChannelZ: what I said or what p3nguin said
19:29.16p3nguinYeah, I know it's harder to type checkinstall as opposed to make install.
19:32.00ChannelZI can always count on your complete lack of sense of humor
19:32.32p3nguinIf that's what you're looking for, I'm your man!
19:32.41esproalright well, now that the module is loading properly, p3nguin do you know of decent documentation to get me started with it? not interested in receiving so much as i am in sending. have something else handling receiving
19:33.27*** join/#asterisk teathsch (~desktop@ip68-4-55-105.pv.oc.cox.net)
19:33.30p3nguinI haven't devised a good method for sending because I don't send faxes very often.  I mainly receive them, and even receiving is not often.
19:33.54p3nguinTake a look at "core show application SendFAX" in your astCLI
19:34.03ChannelZhttp://www.digium.com/en/products/software/faxforasterisk.php#documentation
19:34.38esproChannelZ: that's actually quite terrible :)
19:34.50esproIt's missing at least a couple chapters to actually make it documentation
19:36.13p3nguinSince I have immediate access to dial plan and fax so infrequently, I just hard-code the fax file name into the SendFAX() app.
19:37.00p3nguinWell, that's not completely accurate...
19:37.28p3nguinI code the file name into the macro in the Dial() app, then the macro uses ARG1 to send the fax file.
19:38.07p3nguinIt wouldn't be hard to come up with a plan to make my sending a little more automated, but I just never got around to it.
19:38.51esproSo would it be possible to implement an email to fax gateway?
19:39.07p3nguinThat's what I'll be doing, once I plan it out and make it work.
19:39.45p3nguinI had only one bug with my fax to email, which I finally overcame last night.
19:39.51esprowhat was that?
19:40.38p3nguinAsterisk's System() quit sending emails with mutt for some reason.  After some googling, I found a possible solution of using sudo -u asterisk mutt within System()... and it worked!  I was pleased.
19:41.17*** join/#asterisk darkdrgn2k3 (~darkdrgn2@199.243.221.174)
19:41.21p3nguinIt has been suggested that maybe mutt changed something in its requirements of shell and env that made it quit working.
19:41.43darkdrgn2k3Hey guys, is there any free software out there for jitter test asside from using somethign like ethereal/wireshark
19:42.35p3nguinmtr, maybe?
19:44.45darkdrgn2k3wow i never even new that existed!
19:44.55p3nguinIf I contract with someone else for that person to do work for me, am I a contractor or is the other person doing the work a contractor?
19:45.28jayteeyou're the contractor and the other person is the sub-contractor
19:45.28darkdrgn2k3you are a contractor for your client, but he is a contractor for you..
19:45.34wizbitp3nguin: can i use my VOIP number as a fax number at the same time?
19:45.46darkdrgn2k3wizbit: voip+fax is not the best
19:45.54p3nguinwizbit: If it's SIP and you use g.711, maybe.
19:45.58darkdrgn2k3wizbit: but it is possible. just dont expect perfection unless you use .711
19:46.07wizbiti havent a clue what i use
19:46.11darkdrgn2k3wizbit: otherwise its hit and miss depending on the line..
19:46.22wizbitaye ok
19:46.22Qwellwizbit: find a provider that supports T.38
19:46.33p3nguinMy faxing is over SIP, and it works a huge percentage of the time.
19:46.43darkdrgn2k3as does mine.. but its NOT 100%
19:47.17darkdrgn2k3in my exp.. dont go faster then 14.4 either
19:47.22KattyWHAT UP
19:47.24Kattyasterisk.
19:47.59QwellKatty: Channels be up.
19:48.07Kattywoot for channels up!
19:48.24Kattyalso...i'll up your...channel...in a minute...
19:48.32Kattyerr. nevermind.
19:48.50Kattythat's one of those things that just sounds better in my head.
19:49.12wizbitive taken my 6th call ever since ive installed asterisk 2 years ago
19:49.24p3nguinGreat job!
19:49.27esprowas it a wrong number?
19:49.28wizbit:D
19:49.35wizbitespro: they all were
19:49.38esprolol
19:49.41p3nguinheh
19:49.51Kattythat's hott.
19:51.32darkdrgn2k3when people say "jitter" do they mean avg or wrst?
19:53.38*** join/#asterisk oej (~olle@195.41.130.3)
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19:54.29*** mode/#asterisk [+o BMJ] by ChanServ
19:54.48darkdrgn2k3grrr mtr doesnt show desntination..
19:54.54darkdrgn2k3seems some one is droping icmp packest
19:56.22esprop3nguin: just found this, it's very recent, might be relevant to you as well http://messinet.com/trac/wiki/AsteriskFAXGateway
19:57.14*** join/#asterisk r0d3nt (r0d3nt@foster.stonedcoder.org)
19:57.16p3nguindarkdrgn2k3: avg and worst are just ping times.  To determine jitter, you need to look at the min vs. max over a period of time.
19:57.25darkdrgn2k3aaa
19:57.37darkdrgn2k3so i  could just send a bunch of pings and do the math?
19:57.42Kattyi want pet owl.
19:57.50darkdrgn2k3(since mtr stops at  my ISP becuase they seem to drop icmp packest)
20:00.42p3nguinDropping ICMP is bad.  If you think they are really doing that, call them and tell them to stop it because it makes it hard to conduct normal network operations.
20:00.54darkdrgn2k3well i can only guess
20:01.06darkdrgn2k3becuase i cnat traceroute past my gatway all the way to the destination
20:01.18esproevery time?
20:01.29espro(every host, i mean)
20:01.46p3nguinespro: Unfortunately, I don't see anything there that's useful to me.  I already receive faxes and email them, so I really only need to automate sending of faxes from email.
20:02.16esproit provides both
20:02.28p3nguinBut I already do one, so that rules it out.
20:02.35esprounless it does it better ;)
20:02.44p3nguinI won't be changing my current method to a more complicated one.
20:03.30p3nguinIt doesn't get much easier than receiving the TIFF with ReceiveFAX(), converting to PDF with tiff2pdf, and emailing the PDF with mutt.
20:04.06p3nguinI use msmtpd as my local MTA (as a relay only), and gmail does the actual delivery for me.
20:05.17*** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com)
20:11.57*** part/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu)
20:12.07darkdrgn2k3espro: i get My host, My Router, ISP Routers. ********* Destination
20:13.41esprohow many hops past your gateway?
20:14.40esprojust weird that your isp would be going to the effort of blocking icmp en route but then not blocking it once it's at the destination right? all that's changing is the ttl
20:15.02darkdrgn2k3i take that back.... i dont get ANYTHING at all after the isp's router
20:15.07esproah
20:15.16darkdrgn2k3but i can ping out..
20:16.27p3nguinWhat if you try mtr 205.171.202.203?
20:16.42darkdrgn2k3i get my gateway. isps gateway and ???
20:16.59p3nguinShitty.
20:17.04darkdrgn2k3but pings go throught 64 bytes from 205.171.202.203: icmp_seq=1 ttl=53 time=16.8 ms
20:17.57darkdrgn2k3funny thing is goign the other way. .i get all the hops..
20:18.08p3nguinSo they're blocking traceroute, but allowing ping.  That seems unusual.
20:18.10darkdrgn2k3from my pc -> voip box
20:18.28darkdrgn2k3that traces fine
20:20.04darkdrgn2k3so jitter is max-min
20:20.19darkdrgn2k3LOL 856 ms.. ahahahaa
20:21.06darkdrgn2k3i never understood why pings in the middle are bigger then pings further down...
20:22.39*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
20:22.48p3nguinSpecifically, no that isn't thee jitter amount.
20:23.03darkdrgn2k3so what is considered jitter
20:23.17p3nguinping reports the round-trip time of the ping from you, to the other side, and back to you.
20:23.58darkdrgn2k3so jitter is point a to poitn b not RTT
20:24.12p3nguinJitter is the variation of the delay in one direction.
20:24.33darkdrgn2k3thats what i ment.. max-min of point a to point b
20:24.56darkdrgn2k3so 1/2 of  max/min  of ping is a very poor approximation of jitter
20:25.12p3nguinI guess you could get a rough idea by using half the round-trip times.
20:25.28p3nguinrough is the key word there.
20:25.32darkdrgn2k3yeh
20:25.49darkdrgn2k3other wise i would need to build an app that sends a packet with say a time stamp.. and the other side does the math
20:25.58darkdrgn2k3assuming the clocks are synced
20:26.00p3nguinDo you have access to both sides of the route?
20:26.06darkdrgn2k3yes
20:26.09p3nguinboth ends, rather
20:26.14darkdrgn2k3yes
20:26.54p3nguinI know there's a tool that will measure performance between two points, but I can't remember what it is.  I've used it before, so let me look in my history.
20:27.11darkdrgn2k3i guess the thory would be generate 10 packest exacly 1 second appart
20:27.25darkdrgn2k3on the other side calculate the time differnce between the  packets..
20:27.37darkdrgn2k3take max and min and subtract
20:27.40p3nguiniperf
20:28.01p3nguinspecifically, iperf -u.
20:28.42p3nguinThere are many other options that you can play with.
20:28.50darkdrgn2k3dam not stock in centos
20:29.10p3nguinyum can probably help.
20:29.16darkdrgn2k3sadly no
20:29.20Qwellrpmforge
20:29.29p3nguinOh, so yum _can_ help.
20:29.36darkdrgn2k3hmm i guess so
20:29.40p3nguinSomeone forgot to install rpmforge!
20:29.46darkdrgn2k3LOL
20:29.51darkdrgn2k3at least i got epel installed
20:30.21darkdrgn2k3would help if i spelled it right
20:30.25darkdrgn2k3its in epel..
20:30.28darkdrgn2k3iperf not ipref
20:31.00p3nguinInstalling rpmforge is so much easier today... I remember using dag repositories years ago, and it wasn't as easy as just installing a package locally and then yumming your way to new software.
20:31.21darkdrgn2k3yaathose where the days
20:33.29p3nguinBandwidth 1.05 Mbits/sec  Jitter 0.577 ms
20:33.32darkdrgn2k3[320]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec
20:33.35darkdrgn2k3i dont see jitter
20:33.54p3nguinIt's the one right after bandwidth.
20:34.18darkdrgn2k3nop
20:34.21p3nguinyep
20:34.27p3nguinLook at the server report.
20:34.38darkdrgn2k3somethign is wrong
20:34.39p3nguin[ ID] Interval       Transfer     Bandwidth        Jitter   Lost/Total Datagrams
20:34.52p3nguin[  3]  0.0-10.0 sec  1.25 MBytes  1.05 Mbits/sec   0.577 ms    0/  893 (0%)
20:34.58darkdrgn2k3udp packets are not hitting the server
20:35.10p3nguinfirewalls win again!
20:35.25darkdrgn2k3i holed the firewall
20:35.52p3nguinWhat port did you open for iperf testing?
20:35.58darkdrgn2k35001
20:36.16p3nguinYou're sure you did UDP and not only TCP?
20:36.16darkdrgn2k3iptables -I  INPUT -p UDP --dport 5001  -j ACCEPT
20:36.16darkdrgn2k3.
20:36.33p3nguinI guess so.
20:37.01p3nguinYou're not behind NAT?
20:37.04p3nguinon that side
20:37.12darkdrgn2k3not on that side
20:37.13darkdrgn2k3this side though
20:37.17darkdrgn2k3tcp worked fine btw
20:37.21darkdrgn2k3with -p tcp
20:38.04darkdrgn2k3yeh its the udp packets are are screwed
20:38.21darkdrgn2k3mayb win32 version is screwy?!?!
20:38.37darkdrgn2k3iperf.exe -u -c host
20:40.13darkdrgn2k3tcp works fine -
20:40.13darkdrgn2k3[  4]  0.0-10.3 sec   776 KBytes   619 Kbits/sec
20:42.22*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
20:55.46p3nguinIf you use UDP, the server never sees the traffic?
21:00.20p3nguinIf you'll let me know the host for the iperf server, I'll try it from here.
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21:15.31ChannelZI'd tell you a UDP joke, but you might not get it.
21:15.49QwellI'd tell you a TCP joke, but I already just told you.
21:16.07ChannelZ:P
21:16.18p3nguinI think I like the UDP one best.
21:16.29QwellYou admit that it was clever, or you can gtfo. :p
21:18.46p3nguinYou never acknowledged me on g+, so no.  :(
21:19.15ChannelZCircle denied!
21:19.22p3nguinI mean, you could have told me you were blocking me.
21:20.38ChannelZI should make a circle called "jerk".  Then it'd say things like "Do you wish to add XYZ to your circle jerk?"
21:20.51p3nguin:D
21:21.25p3nguinstays clear of that circle
21:21.42*** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek)
21:22.21*** part/#asterisk mjordan (~mjordan@nat/digium/x-flkhkahiyblfszad)
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21:30.19Qwellp3nguin: It's his circle.  You have no choice.
21:31.12p3nguinI can't deny being put in it?
21:32.49*** join/#asterisk Karen_m (~karen@d50-99-60-236.abhsia.telus.net)
21:33.03Karen_mp3nguin, :)
21:33.23*** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net)
21:34.32p3nguinwaves
21:34.53x1userAnyone who have experience with a2billing, i have few gsm phones connected to asterisk via chan_mobile, but i cant set up a2billing to work?
21:55.06*** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr)
22:10.38treborsuxok i got it
22:10.57treborsuxmy openvox card is working asterisk is aok and freepbx is up
22:11.03treborsuxmy phjones arent here yet
22:11.14treborsuxanyone know where i can get a sip emulator for testing
22:15.04p3nguinMaybe you just need a soft phone.
22:15.09treborsuxcan intel modems be used as fxs or fxo?
22:15.15treborsuxyes i need a soft phone
22:15.15p3nguinnot usually.
22:15.20treborsuxwhere do i get one
22:15.30p3nguinWhat OS are you using?
22:15.52*** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave)
22:15.59treborsuxwindows
22:16.02p3nguinUse zoiper classic.
22:22.47treborsuxjust installed zoiper free
22:24.31p3nguinIt's a good one.
22:28.47treborsuxthanks
22:29.03treborsuxgot to watch videos all day tomorrow and figue out freepbx
22:29.10treborsuxbut i have a system running
22:29.15treborsuxdrivers all ok
22:29.22treborsuxthanks to all
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23:01.31ChannelZDamnit.  PHP doesn't want to run under env
23:08.27*** join/#asterisk matiasjrossi (~matias@host207.190-138-186.telecom.net.ar)
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23:12.46*** part/#asterisk robl^_ (~robl^@pdpc/supporter/active/robl)
23:13.50*** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net)
23:14.14*** join/#asterisk Cain (~Geek@unaffiliated/cain)
23:22.10*** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net)
23:41.22*** join/#asterisk matiasjrossi (~matias@host207.190-138-186.telecom.net.ar)
23:42.21*** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net)
23:43.34michael-iHi everyone. I'm writing a AMI client and had a question regarding the events being fired when connecting a call. Is there any single event packet which will let me know that channelA-and-B have been bridged?
23:44.06michael-iI don't want to maintain a lot of state in the client…but hadn't found what I'm looking for so far. The Bridge and Unlink events fire for other events confusing things.
23:49.58michael-iI guess others have run into this. DTMF key presses result in Unlink and Bridge events: http://forums.digium.com/viewtopic.php?f=1&t=76575&start=0
23:56.10*** join/#asterisk tehrabbitt-1 (~root@unaffiliated/tehrabbitt)
23:56.37tehrabbitt-1Hey, had a quick question... what would cause asterisk to show the wrong CID times?
23:56.53tehrabbitt-1on one phone I see the time at 9:56 (it's really 7:56)
23:56.54*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
23:57.04tehrabbitt-1aand the other phone says 3:57)
23:57.14tehrabbitt-1so one is showing at 9, one is showing at 3
23:57.19tehrabbitt-1and the real time is 7
23:57.46tehrabbitt-1I have the server time set to use ntp / it's set to EST time zone and i've confirmed the system shows the right time by using "date"

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